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Merge in the RTP engine API.

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This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2009-04-02 17:20:52 +00:00
parent 08971ce205
commit 63de834395
24 changed files with 7119 additions and 6130 deletions
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10
apps/app_dial.c
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@@ -54,7 +54,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/rtp.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/cdr.h"
#include "asterisk/manager.h"
#include "asterisk/privacy.h"
@@ -745,7 +745,9 @@ static void do_forward(struct chanlist *o,
char *new_cid_num, *new_cid_name;
struct ast_channel *src;
ast_rtp_make_compatible(c, in, single);
if (single) {
ast_rtp_instance_early_bridge_make_compatible(c, in);
}
if (ast_test_flag64(o, OPT_FORCECLID)) {
new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
new_cid_name = NULL; /* XXX no name ? */
@@ -1745,7 +1747,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
/* Setup outgoing SDP to match incoming one */
ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
if (!outgoing && !rest) {
ast_rtp_instance_early_bridge_make_compatible(tc, chan);
}
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tc);